Gain settings simplified!

This is a discussion on Gain settings simplified! within In-Car Entertainment, part of the Under the Hood category; { Please put this article in the FAQ section }
Hope fully should help most amateurs in tuning the gains ...

Hope fully should help most amateurs in tuning the gains just right....!
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Tweaking an audio system is not as easy as one may think..a few tools, technical know how and definitely very very trained hearing are necessary for it... anyways.. here is the first step towards it.

Gain settings Simplified.

For the ease of it , I have considered a JBL GTO - 75.4 amplifier.

1 - Set everything to "0" - flat eq, hpf off, lpf off bass boost off,loudness off, all x-overs flat... every thing a huge 0 on the HU as well as the amp.

2 - Get the gains all the way down...before you proceed further..

3 - Take a DMM ( digital multi meter ) and a 1KHz test tone to set gains on a class AB amp for mids and highs

4 - Set the DMM to measure AC voltage ( not DC ) , remove ch 3 & 4 speakers as we are going to set only ch1& 2 right now.

5 - Play the 1KHz test tone and put the volume on the HU at 80% of the full volume...

6 - Connect the + of multimeter to +ve of speaker output terminal, and -ve to -ve ....

7 - If the amplifier is 75wrms x 4 ch, you should aim at a voltage reading of 17.3 volts ( this is calculated by ohms law ). So adjust your gain knob till you get a reading of 17.3 volts. this would be a precise technical way to set amplifier gains.

8 - Disconnect ch1&2 and connect 3 & 4 repeat the same process for gain settings of these channels.

9 - slight deviation in these readings would be ok.

================================================== ==How to know what voltage reading to aim for...?

Most simpler way is to use an oscilloscope, check for clipping, you are right play a test track 1Khz at 0dB check for unclipped waveform, this would be your max. volume that you can go up for a clean undistorted input to the amplifier. Now use a 1Khz test signal at -5dB or -10dB for the amplifier tunning again check at the output of the amplifier for unclipped waveform by adjusting the amplifier gain control. use 40kHz test signal for the monoblock amp and 40KhZ at -15dB for amplifier gain adjustment. you are right you need to keep basic seeting in flat range.
Try this, its lot more better and eliminates the use of calculator, pen and paper to do all calculations.

Most simpler way is to use an oscilloscope, check for clipping, you are right play a test track 1Khz at 0dB check for unclipped waveform, this would be your max. volume that you can go up for a clean undistorted input to the amplifier. Now use a 1Khz test signal at -5dB or -10dB for the amplifier tunning again check at the output of the amplifier for unclipped waveform by adjusting the amplifier gain control. use 40kHz test signal for the monoblock amp and 40KhZ at -15dB for amplifier gain adjustment. you are right you need to keep basic seeting in flat range.
Try this, its lot more better and eliminates the use of calculator, pen and paper to do all calculations.

O'scope... agreed... but most DIY'ers cant get hold of an O'scope or RTA for that matter, where as a DMM is easily available and a cheaper instrument too.

With computer based O'scope and RTA programmes, it becomes very necessay to have a calibrated microphone that matches with the sound card of your PC or laptop ....

I couldnt get the sense of using a 40KHz tone ... I mean 40Khz would be entirely out of the human audible spectrum and even most car audio amplifiers to process that tone ...

did you mean 40 Hz ???

also after 1st setting the HU at a volume devoid of clipping with 1KHz 0db, then why would you set the gains on the amp with -5db or -10db tone?? Please elaborate.....!

Sorry typing error I should have typed Headunit in the first line.
Oscilloscope is Just 6 to 7000 only Sir.

I was pointing out about the HU clipping to SADesigns too.

Sirji you are right Oscilloscope are not that costly but taking in account a normal DIYer i think the DMM is best bet out there. For some freaks likes us we will even ABX between Oscilloscope and see which suits our taste better just like Sony or Pioneer woooooofeer

Something I wanna add from my experience:
One should check at what volume the HU clips and set the gains keeping room (considering most people including me still live on 128 - 160 KBPS MP3's).

Eg My HU starts clipping at 30-33 Vol. Max is 35. and i have set gains (according to SADesigns earlier post) at 25 Volume. Now when i play high/good quality music i never had to take it 25 to get the max out of the speakers (for my listening pleasures), whereas when i play some 64Kbp's WMA's (i ripped them long long long time back) i sometimes even have to cross 25 mark. So keeping a head room is always good.

As per my experience, Most Class A, B and A/B amps clip at 70-80% of full potential, HU/s are no exception as they have a class A/B internal amp too..

Like abhibh says ...

He set the gains at 25 when his HU has a max of 35. I would have gone a tad bit more and moved to 28 for setting the gains, ofcourse if there is audible distortion at 28 then would have to move down, but generally 80% of the max would be a good reference point to start at.

About bitrate of MP3's, I would advice using anything above 192kbps...

Although I would still personally prefer Audio CD format, over other compressed formats.

reason being, almost all of the MP3's we convert are from Audio CD format ( 128kbps ) , there is considerable data loss when we convert from this format. until unless the recording studios give us the MP3's directly from the sound track.

I don't recommend the VOM method, unless you know precisely what maximum voltage is provided by your amplifier. If you have a laptop, download winscope. No calibration is really needed, but you have to be careful not to over drive the input of your soundcard. Build a 40:1 voltage divider and you'll be able to connect directly from your amp to your soundcard. If you build an additional 4:1, you can connect from the output of just about any preamp component.

With the scope, you just need to see the shape of the waveform.

Finally, the reason to set gains with a -10 dB tone is because allowing for 10dB of clipping will provide a louder system and a level of distortion you probably won't be able to hear anyway. Distortion on transients has to be nearly 20% before it sounds nasty.

reason being, almost all of the MP3's we convert are from Audio CD format ( 128kbps ) , there is considerable data loss when we convert from this format. until unless the recording studios give us the MP3's directly from the sound track.

That will require pages and pages of explanation.. let me see if i could compress it ...

To understand this, we first need to understand how all the music we listen to is actually recorded in a studio.

A few years back, all this music was recorded on analog magnetic tape, these were usually capable of carrying 24 individual tracks ( meaning 24 individual instruments and sounds not a song track )

Music was known in the multiple of 24 , like 24 /48 / 72 etc, depending on the number of 24 track machines used while recording.

After each track is recorded , the mix was played and recorded on another machine onto a 2 track stereo tape.

Now a days digital recording is taking over, although digital recording depends on the mixing console, computers and softwares... Analog type recording is still an audiophiles favorite as digital recording will sample a sound wave many times per second allowing an illusion of solid sound waves to be created..... in contrast, analog tape captures a sound wave in its entirety, this attributes to the harshness in the higher frequency bandwidth. Where as an analog recording would be clean and warm.

CD quality audio has a sampling rate of 44.1KHz no frequencies above the Nyquist frequency of 22050 Hz are acceptable for recording to avoid aliasing. Very steep sloped LPF are applied to limit the frequencies above 20KHz. This introduces distortion into the audible range.

Uncompressed audio as stored on a cd has a bit rate of 1,411.2 kbps = (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit).... not 128kbps as i had mistakenly written above, excuse me for that.

at this point we also need to know .. Bit depth and Bit rate ...
"Max Lighting" Defines

""Bit depth is the level of data per sample being stored. A recording encoded at 16-bit, means that every sample can store any one of 2^16 (or, 65536) bits per sample.
Bit rate is how many bits of data per second.

Therefore, the bit rate has an upper limit defined by the bit depth.""

So now we know how already a Digitized format stands different from an analog one.

Moving ahead.

MP3.

MP3 in its basics used lossy data compression depending on the algorithm models developed on human perception of hearing.

It is such that it reduces ( deletes/removes) data on the CDAudio.

Imagine stuffing a huge piece of equipment in a smaller box, you would have to dismantle the equipment, try and adjust it inside the box and also may have to omit a few spare parts to shut the box.

MP3 compression is done by removing or reducing certain sections of sound that are beyond the audible spectrum of most humans. This is called perceptual encoding.

Psychoacoustic models are developed to reduce or discard information that would be scarcely audible to human auditory senses.

Different MP3 codecs , encode using different perceptual algorithms. Hence we also notice a difference between different files encoded by different codecs.

Compression efficiency of different encoders are generally defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal.

A 192 kbps compression would give a bigger file, also it would remove lesser data from the given sample as compared to a 128kbps mp3 file and hence the reproduction would be better too.

Again there is CBR and VBR ... constant bitrate and variable bitrate, in CBR the bitrate remains through out the entire piece of data recording, where as in VBR the decoder reduces the bitrate wherever the audio data content is less, meaing in portions of silence or lesses music, the bitrate is reduced. this helps in making the file smaller.

There are some lossless compression codecs too, but these are not entirely lossless formats like the AAC, FLAC , vorbis etc... although they are better compressions than MP3, they give bigger files too.

now use that audio cd to convert into another 128kbps mp3.. do this not more than 5 times... and you will most evidently notice the diminishing quality in sound reproduction on the same audio system.

Bottom line,

Technically and also almost audibly , an audio CD format is the closest to the sound recorded in the studio. ( I said closest, not identical ) ... so further compressions and conversions will definitely lead to data loss and deteriorated quality of reproduction...

While analog does capture sound in its entirety, the ability of infinite number of usage cycles and perfect replication capabilities make digital a formidable way forward.

While 44.1kHz/16 bits per sample is enough for us mortals, for the audiophiles and studio work, 192kHz sampling with 24 bits per sample depths are available. Hell, it is nearly accessible to everyone with standards like DVD-Audio and the HD formats. We just have to wait for such things to trickle down to our part of world.

As per my experience, Most Class A, B and A/B amps clip at 70-80% of full potential,

About bitrate of MP3's, I would advice using anything above 192kbps...

reason being, almost all of the MP3's we convert are from Audio CD format ( 128kbps ) , there is considerable data loss when we convert from this format. until unless the recording studios give us the MP3's directly from the sound track.

Quote:

Originally Posted by gunbir

Can you please elaborate on this? I find this most interesting...

Fenil,

If I were to a rip an MP3 (under atleast 256k) from a Commercial CD as against ripping an MP3 from the Studio Master I would exect both MP3 to sound just about the same. The limitation being the quality of the MP3 encoding algorithim. So even if you used the original master I dont see how the MP3 quality wold be better than that of an MP3 ripped from a well recorded Commercial CD.

Why do you not prefer MP3s above 192kbps. Most of my recordings are at bitrates (I use VBR so I dont get a consistent bit rate) exceeding 192kbps using EAC and LAME.

Quote:

Originally Posted by SADesigns

Now a days digital recording is taking over

Take your favourite track ( on an original audio cd )
convert it using lame decoded into 128kbps mps file. ( note )
now burn that mp3 on your computer, onto an audio cd format again.
now use that audio cd to convert into another 128kbps mp3..

Technically and also almost audibly , an audio CD format is the closest to the sound recorded in the studio.

Going from MP3 to CDA makes little sense even if it high bit rate (read as 256k +) MP3. Why would one do that?

While it is true that some well recorded and produced LPs on a very good 'table can sound better than the CD version I dont see 2 channel home audio tape being a viable alternate because by design the medium has significant compression. 1" tape running at 15ips is a different ball game but this is not a true home audio alternate for exmaple we cant compare a (circa '80s) Sony 3324 to a $199-$399 CD Player.

Tape is also suseptable to drop outs. I used to use a PCM encoder with a VCR in the late '80s for mixes (what we in the post "ipot" world call playlists) and dropouts on PCM sound ugh!

Quote:

Originally Posted by ImmortalZ

192kHz sampling with 24 bits per sample depths are available. We just have to wait for such things to trickle down to our part of world.

fenil sir Thank you for the clear explanation of gains. it was truly helpful and has let me revisit some forgotten things.

Since this has got to an MP3 discussion also let me also elaborate that sound cards capable of 48000hz, 24-bits at rates of upto 320kbps have been around since the 90's.

@navin sir i think he said we SHOULD use a file encoded 192kbps and ABOVE.

@ALL coming back to the discussion. i have personally been encoding to 44.1 24-bit CBR 320kbps files since 99. today we have gotten to encoding in 5.1 channel mode with just over 120kbps per channel at 44.1 16-bit which is more that enough audio information to confuse the mind in terms of quality.

Also please note that the new mp4 codecs and higher systems along with the new divx formats are essentially only dropping out masked elements in audio. things that you basically cannot actually hear and even today close to 50% of that bitstream is audio above 5khz. (source from the net. i may be wrong.)